High frequency signal construction method and apparatus

ABSTRACT

A method of adding high frequency content to an input signal to form an augmented signal, the method comprising the steps of: (a) providing an initial signal having a first predetermined lower spectral range; (b) utilising the initial signal to form synthesised high frequency components of the initial signal; (c) filtering the initial signal with a low pass filter and filtering the synthesised high frequency components with a high pass filter (d) combining the filtered signals to form the augmented signal.

BACKGROUND

[0001] 1. Field of the Invention

[0002] The present invention relates to the synthesis of high frequencysignals and, in particular, discloses a method and system forsynthesising high frequency audio signals.

[0003] 2. Background of the Invention

[0004] The digital recording of audio signals has become extremelypopular. The most popular format for recording is the CD audio formatwhich samples a signal at approximately 44.1 KHz. This is likely toproduce a corresponding audio range of approximately 20 kHz which wasthought to be adequate for reproducing the audio range that the humanear can detect. However, it is thought by some that the human ear isable to colour an audio signal through the utilization of portions of asignal above 20 kHz. Hence, recent standards have proposed either an88.2 or a 96 kHz sampling rate. There is therefore the significantproblem of how one takes, for example, a 44.1 kHz recorded signal andremasters the signal in say an 88.2 kHz format. One standard techniqueutilised is to use an interpolator that also uses some kind of linearfilter to perform an anti alias filtering operation.

[0005] For the purposes of further discussion, the following terminologyis defined:

[0006] The original signal is called the Original Audio Signal.

[0007] The original audio sample rate is called the Original SampleRate.

[0008] The original audio signal is believed to be “accurate” up to afrequency known as the Original Frequency Range.

[0009] The Original Half Nyquist Frequency is defined as 0.5 times theOriginal Sample Rate.

[0010] The interpolated signal is called the Interpolated Audio Signal.

[0011] The new (higher) audio sample rate is called the InterpolatedSample Rate.

[0012] The Interpolated Half Nyquist Frequency is defined as 0.5 timesthe Interpolated Sample Rate.

[0013] The Oversampling Ratio is the Interpolated Sample Rate divided bythe Original Sample Rate.

[0014] Typical values of the above defined quantities are:

[0015] For a CD player with 4× oversampling D/A converters

[0016] Original Sample Rate=44,100 Hz

[0017] Original Frequency Range=20,000 Hz

[0018] Original Half Nyquist Frequency=22,050 Hz

[0019] Interpolated Sample Rate=176, 400 Hz

[0020] Interpolated Half Nyquist Frequency=88, 200 Hz

[0021] In a system like this, the Original Audio Signal only containsreliable content up to 20 kHz, but it is assumed it may be desirable tosynthesize new high frequency content up to say 88.2 kHz.

[0022] For a DVD player with 2× oversampling D/A converters:

[0023] Original Sample Rate=48,000 Hz ‘Original Frequency Range=20,000Hz Original Half Nyquist Frequency=24,000 Hz

[0024] Interpolated Sample Rate=96,000 Hz Interpolated Half NyquistFrequency=48,000 Hz

[0025] In a system like this, the Original Audio Signal only containsreliable content up to 20 kHz, but it may be desirable to synthesize newhigh frequency content up to 48 kHz. [0024] The standard prior artanti-aliasing approach to higher sampling rate extension operates on theprinciple that as no information about what audio content may haveexisted above the Original Half Nyquist Frequency is provided in theoriginal audio material, it is necessary to ensure that an InterpolatedAudio Signal has zero content in this upper frequency range.

[0026] The standard prior art method for producing an interpolatedsignal will now be described. Turning initially to FIG. 1, an originalaudio signal 1 is provided having samples e.g., 11,12. The samples areassumed to have been provided at a standard rate. The first step informing the interpolated signal is to zero pad the audio signal asillustrated in FIG. 2. In zero padding, zero value signals e.g., 14,15are added to the signal between samples. Next, as illustrated in FIG. 3,an interpolation process is provided where the signal e.g., 18 is formedfrom an interpolation of the two signals 17,19. In the example provided,the interpolated sample rate is twice the original sample rate and hencethe over sampling ratio is 2 with one zero sample inserted between eachsample of the original audio signal. The zero-padding technique resultsin aliasing, meaning that the low frequency audio signal is duplicatedin higher frequency bands. These higher frequency replicas (calledaliases) are then filtered out (using a low-pass filter), to leave theInterpolated Audio Signal.

[0027] An example of aliasing is illustrated in FIG. 4 where an originalaudio signal having a frequency spectrum 21 is zero padded resulting inthe zero padded audio signal having a frequency spectrum 23,24 with thelower frequency being replicated in high frequency bands. Theinterpolation process is equivalent to applying a low-pass filter 27which results in the interpolated audio signal 29 which substantiallyreflects the original audio signal 21.

[0028] The arrangement of the prior art has a significant disadvantagein that none of the high frequency spectrum is utilized when are-sampling occurs.

SUMMARY

[0029] It is an object of the present invention is to provide foralternative forms of high frequency signal extension of signals.

[0030] According to a first aspect of the invention there is providedmethod of adding high frequency content to an input signal to form anaugmented signal, the method comprising the steps of:

[0031] (a) providing an initial signal having a first predeterminedlower spectral range;

[0032] (b) utilising said initial signal to form synthesised highfrequency components of said initial signal which extend beyond saidlower spectral range;

[0033] (c) filtering said initial signal with a low pass filter andfiltering said synthesised high frequency components with a high passfilter

[0034] (d) combining said high and low pass filtered signals to formsaid augmented signal.

[0035] Preferably, step (b) further comprises:

[0036] (i) for at least one portion of the input signal, determining thespectral content of said portion;

[0037] (ii) extrapolating a high frequency end portion of the spectralcontent to form said synthesised higher frequency components of saidsignal.

[0038] Conveniently, said portion is multiplied with a window functionprior to determination of the spectral content and said synthesisedhigher frequency components are summed in an overlap-add fashion.

[0039] Advantageously, the method includes the step of dividing theinput signal into a plurality of overlapping blocks, with each blockbeing multiplied by a sliding window function to yield a series ofwindowed portions from which high frequency components are successivelysynthesised.

[0040] The window may be of a Gaussian or Hanning form.

[0041] The invention extends to a method of adding high frequencycontent to an input signal to form an augmented signal, the methodcomprising the steps of:

[0042] (a) providing an initial signal having a first predeterminedlower spectral range;

[0043] (b) utilising said initial signal to form synthesised highfrequency components of said initial signal which extend beyond saidlower spectral range, wherein step (b) further comprises:

[0044] (i) for at least one portion of the input signal, determining thespectral range of said portion;

[0045] (ii) extrapolating a high frequency end portion of the spectralrange to form said synthesised higher frequency components of saidsignal.

[0046] Conveniently, said portion is multiplied with a window functionprior to determination of the spectral content, and said synthesisedhigh frequency components are summed in an overlap-add fashion.

[0047] Preferably, the method includes the steps of dividing the inputsignal into a plurality of overlapping blocks, with each block beingmultiplied by a sliding window function to yield a series of windowedportions from which high frequency components are successivelysynthesised.

[0048] Conveniently, at least some of the highest frequency componentsof said spectral content are discarded prior to the extrapolation of theremaining high frequency components.

[0049] Typically, the step of extrapolating said high frequency endportion comprises the steps of sampling the high frequency components,defining an extrapolation factor based on a geometric progression, andgenerating said geometric progression on the basis of the sampled highfrequency components.

[0050] According to still further aspect of the invention there isprovided apparatus for adding high frequency content to an input signalto form an augmented signal, the apparatus comprising:

[0051] (a) a synthesising processor for synthesising high frequencycomponents from an initial signal having a first predetermined lowerspectral range, said high frequency components extending beyond saidlower spectral range;

[0052] (b) a low pass filter for filtering said initial signal;

[0053] (c) a high pass filter for filtering said synthesised highfrequency components;

[0054] (d) a combiner for combining said high and low pass filteredsignals to form said augmented signal.

[0055] Preferably, said synthesising processor comprises means fordetermining the spectral content of at least one portion of said inputsignal and means for extrapolating from a high frequency end portion ofsaid spectral content to form said synthesised high frequency componentsof said signal.

[0056] The invention still further provides an apparatus for adding highfrequency content to an input signal to form an augmented signal, theapparatus comprising:

[0057] (a) a synthesising processor for synthesising high frequencycomponents from an initial signal having a first predetermined lowerspectral range, said high frequency components extending beyond saidlower spectral range;

[0058] (b) means for dividing the input signal into a plurality ofoverlapping portions;

[0059] (c) means for determining the spectral content of each of saidoverlapping portions; (d) means for extrapolating the high frequency endportion of the spectral content to form said synthesised high frequencycomponents of said signal; (e) means for summing said synthesised highfrequency components in a overlap-add fashion.

BRIEF DESCRIPTION OF THE DRAWINGS

[0060] Notwithstanding any other forms which may fall within the scopeof the present invention, preferred forms of the invention will now bedescribed, by way of example only, with reference to the accompanyingdrawings in which:

[0061]FIG. 1 illustrates a sampled original audio signal;

[0062]FIG. 2 illustrates a zero padded audio signal;

[0063]FIG. 3 illustrates an interpolated audio signal;

[0064]FIG. 4 illustrates the prior art process of forming aninterpolated audio signal in the frequency domain;

[0065]FIG. 5 illustrates the basic process of forming an augmented audiosignal in accordance with one embodiment;

[0066]FIG. 6 illustrates the frequency extension process of FIG. 5;

[0067]FIG. 7 illustrates a first embodiment of an apparatus forgenerating an augmented audio signal; and

[0068]FIG. 8 illustrates a second embodiment of an apparatus forgenerating an augmented audio signal.

DESCRIPTION OF PREFERRED AND OTHER EMBODIMENTS

[0069] In the preferred embodiment, there are provided varioustechniques for creating a reasonable estimate of the frequency responseof the audio signal above the original frequency range. Further, thetechniques are extended to include techniques for incorporating theextended frequency response signal into the interpolated audio signalwhilst ensuring that the interpolated audio signal is an accurate matchto the original signal in the more important lower frequency range.

[0070] All frequency extension techniques are, by definition,non-linear, because they cause the creation of new frequency content inthe output signal that was not present in the input signal. Hence it isextremely difficult to ensure that a Frequency Extension Technique doesnot also introduce non-linear/distortion artifacts that are audiblewithin the Original Frequency Range of the Original Audio Signal. Hence,the preferred embodiment proposes that the information from the OriginalAudio Signal, within the Original Frequency Range, should be preserved,by reinserting it into the Interpolated Audio Signal.

[0071] An example of this arrangement is shown at 30 in FIG. 5 where anoriginal audio signal having a spectrum 21 is adapted utilizing afrequency extension technique 36 which is described in more detail belowso as to provide for an extended audio signal having an extendedfrequency 32. The original signal is low-pass filtered using a low passfilter 37 and the extended audio signal is high-pass filtered using ahigh pass filter 38 before they are combined at 39 to produce theinterpolated or augmented audio signal 34,35 which extends into the highfrequency range. Hence, the Interpolated Audio Signal is composed of twosignal components added together:

[0072] I. The low frequency part of the Original Audio Signal.

[0073] II. The high frequency part of the Extended Audio Signal.

[0074] In many cases, the Extended Audio Signal will be a very closeapproximation to the Interpolated Audio Signal, but the use of thelow-pass 37 and high-pass 38 filters, and the summing element 39, ensurethat any inaccuracies in the low frequency part of the Extended AudioSignal are removed, and replaced with the more accurate low-frequencycomponents from the Original Audio Signal.

[0075] The use of the low-pass/high-pass technique, has the followingbenefits:

[0076] I. Low frequency information (from the Original Audio Signal) ispreserved in an unaltered form;

[0077] II. High frequency information approximating the likely extensionof the Original Audio Signal is added without affecting the lowfrequency information, but still using the Original Audio signal as abasis for the extension.

[0078] One method for high frequency extension 36 can operate by workingon a sliding window on the Original Audio Signal, so that, with eachiteration of the process, a windowed segment of the Original AudioSignal is analyzed, say in the Fourier domain, resulting in the methodas is shown in FIG. 6, which shows a single iteration.

[0079] A segment or block 42.1 of the original audio signal 42 ismultiplied with a window 43 (which can be of Gaussian form). Themultiplied result of the two signals is then transferred into theFourier domain using a Fast Fourier Transform (FFT) 44 or the like so asto produce a frequency response 41. The frequency response curve 41 willoften include a ringing peak 46 that appears due to the anti-aliasfilter. However, the lower frequency points 47 and 48 adjacent the peakbut still at the high frequency end of the response can be relied uponas truer indicators of the high frequency content of the original audiosignal. Hence an extrapolation process 50 can be carried out so as toextend the representative high frequency components 47 and 48 of theaudio signal. The components at points 47 and 48 are extrapolated,thereby yielding a reasonable estimate of the extended audio signal 49.The extended audio signal 49 then undergoes an inverse fast Fouriertransform 60 before being multiplied by a Gaussian window 51 to yield apartially computed output audio signal 52 in the time domain.

[0080] Obviously, various other extrapolation techniques can beutilized. For example, in a 32 tap FFT filter, the FFT bins ranging from13.5,15,16.5,18 to 19.5 kHz can be used. The 19.5 kHz bin may beadversely influenced by the peak 46. One form of extension can be madeby extrapolating the difference between the samples 47 and 48corresponding, say, to the 16.5 and 18 kHz bins, to higher frequencies,and by continuing them in a geometric series, as outlined in themathematical summary below.

[0081] A high frequency audio signal augmentation system of the typeschematically illustrated in FIG. 7 can be utilized.

[0082] Let x(k) be the original input signal and y(l), the ExtrapolatedSignal to be created.

[0083] The oversampling ratio can be S (typically, S=2 or S=4). This isimplemented using an oversampler 54, into which the original audiosignal is inputted.

[0084] Let the original FFT length be N; hence, the extended FFT length,N′=N.S

[0085] Let the two extrapolation FFT bins into which the representativehigh frequency samples are loaded (say 16.5 and 18 kHz) be defined as e₁and e₂

[0086] Let the Overlap be L

[0087] Process the input signal blocks in overlapping blocks as follows,using the segmenting function 56. Each of the overlapping blocks aresuccessively multiplied with a Gaussian window function 43 usingmultiplier 58. The forward shift in the window function essentiallydefines the blocks and their degree of overlap.

[0088] In iteration p:

[0089] Take the windowed input block of length N b_(p)(i)=x(L.p+I).w(i)for i=0 . . . N−1

[0090] Take the (real) FFT of this input block B_(p)=FFT[b_(p)] usingFFT processor 44 (note, b_(p) is of length N, but B_(p) is of lengthN/2+1, because we are using the real FFT)

[0091] Define the Extrapolation factor f that tells us how the frequencyresponse of B_(p) can be extrapolated beyond the FFT bins e₁ and e₂, bya geometric progression. The Extrapolation factor f is a complex number,constrained to lie on or within the unit circle:

f=B _(p)(e ₂)/B _(p)(e ₁) if |B _(p)(e ₂ |<|B _(p)(e ₁)|

f=0 if B _(p)(e ₁)=0

f=(B _(p)(e ₂)/B _(p)(e ₁))/|B _(p)(e ₂)/B _(p)(e ₁) otherwise

[0092] This is achieved using a frequency extrapolation processor 50.

[0093] Form the new, extended frequency response B′_(p) (which is oflength N′/2+1) defined as:

B′ _(p)(i)=B _(p)(i) (0≦i≦e ₁)

B′ _(p)(i)=B _(p)(e ₁).f ^(j−e1)(e ₁ <i≦N′/2+1)

[0094] Transform this extended frequency response back to the timedomain using Inverse FFT processor 60, creating a time-domain signalblock of length N′:

b′_(p)=IFFT{B′_(p)}

[0095] This block of output is then summed into an output buffer afterapplying a suitable window w′ which can be in the form of the Gaussianwindow 51. The summer and buffer are shown at 62. Many other differentwindow functions can be used with this method, with one desirable windowincluding a Hanning window.

y _(p)(S.L.p+i)=y _(p−1)(S.L.p+i)+b′ _(p)(i).w′(i) for i=0 . . . N′−1

[0096] Following this summation operation, the first S.L samples areavailable to be output:

y(S.L.p+i)=yp(S.L.p+i) for i=0 . . . S.L−1

[0097] Part of the oversampled input audio signal is low pass filteredusing the low pass filter 37 having a cut-off frequency of 19 kHz,towards the end of the audible frequency range. The summed extrapolatedtime domain samples are high pass filtered using the high pass filter38, which has a cut-off frequency of 19 kHz, matching that the low passfilter 37 so as to prevent overlap of the low (audible) and high(ultrasonic) frequency parts of the signal, thereby to prevent lowerfrequency components of the ultrasonic signal interfering with thehigher frequency components of the audible signal. These are then summedat summer 39 to yield an extrapolated or augmented output audio signal.

[0098] Referring now to FIG. 8, a further embodiment of a signalaugmenting apparatus is shown which is specific to a doubling of thesampling rate. One part of the audio signal is processed through asample rate converter or oversampler 65 which includes a 19 kHz low passfilter, with the low frequency output portion being fed to the summer39. The other part of the input audio signal is fed to a segmenter 67where it is broken into overlapping blocks. Each block is half as longas the equivalent blocks in FIG. 7 for the reason that the audio datahas not been sample-rate converted or oversampled. Each block is in turnmultiplied with the Gaussian window 43 and the result is converted intothe frequency domain using a fast Fourier transform function 66 oflength N, half that of the function 44 in FIG. 7. The frequency responseis then processed using frequency extrapolator or interpolator 68,resulting in an extended audio signal of the type illustrated at 49 inFIG. 6. The augmented signal is inverse fast Fourier transformed backinto the time domain using an inverse fast Fourier transform function60. The resultant time domain signal has a length 2N. From then on, theprocessor is identical to that illustrated in FIG. 7.

[0099] It will be appreciated that in both FIGS. 7 and 8 each successiveblock defined by a shift in the window undergoes the process illustratedin FIG. 6 to yield a succession of partially computed output audiosignals 52 which are then buffered and summed before being high passfiltered at 38. The main difference is that in FIG. 8 the frequencyextrapolator 68 both doubles the length of the shorter frequency vectorat the same time as extrapolating it.

[0100] In certain forms of the invention, the high and low pass filtersmay be eliminated. By way of example, with reference to FIG. 7 oneembodiment may exclude low pass filter 37, high pass filter 38 andsummer 39, with the extrapolation technique being sufficient to avoidcorruption of the audible frequencies.

[0101] In a further possible embodiment, blocks 34, 37, 38 and 39 may beincluded, and the remaining blocks may be replaced by a crudeinterpolation function. In this embodiment, the high and low passfilters serve the primary function of preventing the high frequencyportion of the signal from corrupting the low frequency portion.

[0102] The invention has numerous audio recordal and playbackapplications, including the following:

[0103] remastering of digital and analogue recordings having arelatively low sampled rate in the region of 44 kHz;

[0104] processing of recorded audio signals in CD, DVD and similarplayers having oversampling functions, where the audio content iscoloured with but not altered by ultrasonic components;

[0105] particular application in audio playback devices such as CD andDVD players, and in any similar devices where oversampling is utilized.

[0106] It would be appreciated by a person skilled in the art thatnumerous variations and/or modifications may be made to the presentinvention as shown in the specific embodiment without departing from thespirit or scope of the invention as broadly described. The presentembodiment is, therefore, to be considered in all respects to beillustrative and not restrictive.

1. A method of adding high frequency content to an input signal to forman augmented signal, the method comprising the steps of: (a) providingan initial signal having a first predetermined lower spectral range; (b)utilising said initial signal to form synthesised high frequencycomponents of said initial signal which extend beyond said lowerspectral range; (c) filtering said initial signal with a low pass filterand filtering said synthesised high frequency components with a highpass filter (d) combining said high and low pass filtered signals toform said augmented signal.
 2. A method as claimed in claim 1 whereinsaid step (b) further comprises: (i) for at least one portion of theinput signal, determining the spectral content of said portion; (ii)extrapolating a high frequency end portion of the spectral content toform said synthesised higher frequency components of said signal.
 3. Amethod as claimed in claim 2 wherein said portion is multiplied with awindow function prior to determination of the spectral content and saidsynthesised higher frequency components are summed in an overlap-addfashion.
 4. A method as claimed in claim 3 which includes the step ofdividing the input signal into a plurality of overlapping blocks, witheach block being multiplied by a sliding window function to yield aseries of windowed portions from which high frequency components aresuccessively synthesised.
 5. A method as claimed in claims 3 or 4wherein said window is of a Gaussian or Hanning form.
 6. A method ofadding high frequency content to an input signal to form an augmentedsignal, the method comprising the steps of: (a) providing an initialsignal having a first predetermined lower spectral range; (b) utilisingsaid initial signal to form synthesised high frequency components ofsaid initial signal which extend beyond said lower spectral range,wherein step (b) further comprises: (i) for at least one portion of theinput signal, determining the spectral range of said portion; (ii)extrapolating a high frequency end portion of the spectral range to formsaid synthesised higher frequency components of said signal.
 7. A methodaccording to claim 6 wherein said portion is multiplied with a windowfunction prior to determination of the spectral content, and saidsynthesised high frequency components are summed in an overlap-addfashion.
 8. A method according to claim 7 which includes the steps ofdividing the input signal into a plurality of overlapping blocks, witheach block being multiplied by a sliding window function to yield aseries of windowed portions from which high frequency components aresuccessively synthesised.
 9. A method as claimed in any one of claims 2to 8 wherein at least some of the highest frequency components of saidspectral content are discarded prior to the extrapolation of theremaining high frequency components.
 10. A method as claimed in any oneof claims 6 to 9 in which the step of extrapolating said high frequencyend portion comprises the steps of sampling the high frequencycomponents, defining an extrapolation factor based on a geometricprogression, and generating said geometric progression on the basis ofthe sampled high frequency components.
 11. A method as claimed in anyprevious claim wherein said signal is an audio signal, said lowerspectral range corresponds to an audible component of said signal andsaid high frequency components correspond to ultrasonic components. 12.A method of adding high frequency content to an input signalsubstantially as hereinbefore described with reference to FIGS. 5 to 8of the accompanying drawings.
 13. Apparatus for adding high frequencycontent to an input signal to form an augmented signal, the apparatuscomprising: (a) a synthesising processor for synthesising high frequencycomponents from an initial signal having a first predetermined lowerspectral range, said high frequency components extending beyond saidlower spectral range; (b) a low pass filter for filtering said initialsignal; (c) a high pass filter for filtering said synthesised highfrequency components; (d) a combiner for combining said high and lowpass filtered signals to form said augmented signal.
 14. An apparatusaccording to claim 13 in which said synthesising processor comprisesmeans for determining the spectral content of at least one portion ofsaid input signal and means for extrapolating from a high frequency endportion of said spectral content to form said synthesised high frequencycomponents of said signal.
 15. An apparatus for adding high frequencycontent to an input signal to form an augmented signal, the apparatuscomprising: (a) a synthesising processor for synthesising high frequencycomponents from an initial signal having a first predetermined lowerspectral range, said high frequency components extending beyond saidlower spectral range; (b) means for dividing the input signal into aplurality of overlapping portions; (c) means for determining thespectral content of each of said overlapping portions; (d) means forextrapolating the high frequency end portion of the spectral content toform said synthesised high frequency components of said signal; (e)means for summing said synthesised high frequency components in aoverlap-add fashion.
 16. Apparatus for adding high frequency content toan input signal substantially as herein before described with referenceto FIGS. 5 to 8 of the accompanying drawings.